Re: Doubt about Caller ID

wilsonsant · ‎04-19-2018

Hi Guys,

My Customer have the solution with CUCM 10.5.2 and 06 sites running SIP protocol. The Customer changed the protocol, previouly were H.323 and now it running SIP with change range Phone number. In any sites the Caller ID that is appearing is the old number. The Carrier informed that forward the number that my network send for They. My question is: Where I see in the gateway what number is send for the Carrier.

Thanks,

Wilson

Maren Mahoney · ‎04-19-2018

debug ccsip messages will show you the signaling exchange between CUCM and your router, and your router and the carrier.

You mention that you have changed phone numbers with the change in carrier. Did you reconfigure the External Phone Number Mask on the Phone-Dn for your users? (And/Or transformation patterns if you are doing Globalized Call Routing?)

wilsonsant · ‎04-20-2018

Hi Marren,

Thanks a lot for Your contact. Unfortunatelly my profile not allow run debug.

About External Phone Number Mask are blank for all extensions.

About transformation patterns if you are doing Globalized Call Routing. The answer is yes

Thank You,

Regards,

Wilson

Maren Mahoney · ‎04-20-2018

OK. Since you are doing Globalized Call Routing, have the transformation patterns used by the SIP Trunk been adjusted to reflect the new DIDs with the new carrier?

These would be the Calling Party Transformation patterns indicated in the Outbound Call Routing section of your SIP Trunk. On the trunk, look at Outbound Cal Routing > Calling Party Transformation CSS. Backtrack and figure out which set of transformation patterns are in that CSS. Are they set to the new DNs?

And, yes, please also post the config of your router. It may be that the router is doing the old-numbering digit manipulation.

Jaca · ‎04-19-2018

You might be able to use debug isdn q931

Add a debug condition to limit the output if you use it

Something like:

voicegate# term mon

voicegate# debug condition called 7895551212

voicegate# debug isdn q931

after the capture:

voicegate# term no mon

voicegate# u all

The called number can be your cell phone.

You can also look at RTMT logs and CDR from CM, but instant information will come from the debug.

The debug ccsip messages will work even better if you don't have a trunk to the PSTN.

Here is a Cisco Command Reference guide for SIP debug conditions and output filtering:

http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_--_SIP_Debug_Output_Filtering_Support

wilsonsant · ‎04-20-2018

Hi Jaca,

Thanks a lot for Your contact. Unfortunatelly my profile don´t have permission to run debug.

About RTMT what logs that I set will show this?

Very usual the link that You send. Thank You

Regards,

Wilson

Jaca · ‎04-20-2018

Once you have RTMT up and running go to:

Voice/Video || Session Trace Log View || Real Time Data

Add the number called and the date and time.

I see from your other posts you have no CLID set within CM. You may have a translation within the gateway sending outbound, but with no running configuration I'm just guessing.

wilsonsant · ‎04-20-2018

Hi Jaca,

About RTMT that You mentioned: Voice/Video || Session Trace Log View || Real Time Data, it show the calling number - originate number, but, the when called receive that the calling number show is different.

About translation between CUCM and Gateway, in the CUCM there are not this translation. As I see in the gateway?

Thanks,

Wilson

Jaca · ‎04-20-2018

Post the configuration of the gateway that holds the trunk in question please.

R0g22 · ‎04-20-2018

Post your router config. Do a DNA on CUCM for an outbound number so that we can see what is the CLID being extended. That way we can tie it to any potential translations on the GW that might be taking affect since you cannot take a debug.

wilsonsant · ‎04-20-2018

Hi Nipun,

Thanks a lot for Your contact. Follow the gateway config

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2018.04.20 16:11:34 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...

Current configuration : 14688 bytes
!
! Last configuration change at 19:00:54 GMT Thu Apr 19 2018 by us_bdavidson
!
version 15.4
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
!
hostname XPTO_BR500_VGW
!
boot-start-marker
boot system flash c2951-universalk9-mz.SPA.154-3.M5.bin
boot-end-marker
!
!
! card type command needed for slot/vwic-slot 0/1
logging queue-limit 10000
logging buffered 10000000
logging rate-limit 10000
enable secret 5 $1$0/Aw$7X8JB2hhHEqmtI/UcdUNW.
!
aaa new-model
!
!
aaa authentication login default group tacacs+ local
aaa authentication enable default group tacacs+ enable
aaa authentication ppp default local
aaa authorization config-commands
aaa authorization exec default group tacacs+ local if-authenticated
aaa authorization commands 15 default group tacacs+ none
aaa accounting exec default start-stop group tacacs+
aaa accounting commands 15 default stop-only group tacacs+
!
!
!
!
!
aaa session-id common
clock timezone GMT -3 0
!
!
!
!
!
!
no ip source-route
!
!
!
!
!
!
!
!
no ip bootp server
no ip domain lookup
ip domain name XPTO.com
ip cef
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
!
trunk group PSTN
hunt-scheme round-robin both up
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
!
cts logging verbose
!

voice-card 0
dspfarm
dsp services dspfarm
!
!
voice call send-alert
voice rtp send-recv
!
voice service pots
fax rate disable
!
voice service voip
ip address trusted list
ipv4 192.168.104.2
ipv4 192.168.104.1
ipv4 10.64.0.8
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
modem passthrough nse codec g711ulaw
sip
session transport tcp
min-se 2400 session-expires 2400
registrar server expires max 600 min 60
asserted-id pai
midcall-signaling passthru
g729 annexb-all
sip-profiles 1
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
!
voice class sip-profiles 1
request ANY sdp-header Audio-Attribute modify "a=inactive" "a=sendrecv"
request REINVITE sdp-header Audio-Attribute modify "a=sendonly" "a=sendrecv"
request ANY sdp-header Audio-Attribute modify "a=sendonly" "a=sendrecv"
!
!
voice class e164-pattern-map 1
description CARRIER - BR500 DID Ranges
e164 +337338344704
e164 +6622723095..
e164 +6622424285[0-4].
e164 +6622681697[4-9].
!
!
voice class e164-pattern-map 2
description CARRIER - BRA DID Ranges
e164 723095..
e164 681697[4-9].
e164 85[0-4].
e164 95..
e164 97[4-9].
!
!
!
voice register global
mode srst
timeouts interdigit 5
system message WAN Outage - Fallback Mode
max-pool 250
!
voice register pool 1
translation-profile incoming SRST-IN
id network 10.0.0.0 mask 233.0.0.0
dtmf-relay rtp-nte sip-notify
voice-class codec 1
no vad
!
!
!
voice translation-rule 10
rule 1 /^$[2-5].......$$/ /+6622\1/
rule 2 /^$[79][6-9].......$$/ /+6622\1/
rule 3 /^$[1-9][1-9][2-9].......$$/ /+33\1/
rule 4 /^$[1-37-9][1-9][79]........$$/ /+33\1/
rule 5 /^\00$.*$$/ /+\1/
rule 6 /^$.*$$/ /+\1/
!
voice translation-rule 20
rule 1 /^$97..$$/ /+66226816\1/
rule 2 /^$8...$$/ /+66224242\1/
rule 3 /^$68169...$$/ /+6622\1/
rule 4 /^$95..$$/ /+66227230\1/
rule 5 /^$723095..$$/ /+6622\1/
!
voice translation-rule 30
rule 2 /.*/ /8500/
rule 3 /^\+.*/ /8500/
!
voice translation-rule 40
rule 2 /^0$[2-9].......$$/ /\1/
rule 3 /^0$[79]........$$/ /\1/
rule 4 /^0$19.$$/ /\1/
rule 5 /^0$181$$/ /\1/
rule 6 /^0$911$$/ /\1/
rule 7 /^0$112$$/ /\1/
rule 8 /^00$[1-9][1-9][2-9].......$$/ /076\1/
rule 9 /^00$[1-9][1-9][2-9]........$$/ /076\1/
rule 10 /^00$[4-6][1-9]9[6-9].......$$/ /076\1/
rule 11 /^00$[1-37-9][1-9][79]........$$/ /076\1/
rule 12 /^00$[38]00.......$$/ /0\1/
rule 13 /^000$[1-9].*$$/ /0076\1/
rule 14 /^0$.*$$/ /\1/
!
voice translation-rule 50
rule 1 /^$97[4-9].$$/ /+66226816\1/
rule 2 /^$329.$$/ /+33335284\1/
rule 3 /^$8[78]..$$/ /+33338526\1/
rule 4 /^0$/ /+662268169740/
!
voice translation-rule 60
rule 11 /^\+6622$[2-9].......$/ /0\1/
rule 12 /^\+6622$[79]........$/ /0\1/
rule 13 /^\+33$[1-9][1-9][2-9].......$/ /00\1/
rule 14 /^\+33$[1-37-9][1-9][79]........$/ /00\1/
rule 15 /^\+$[1-9].*$/ /000\1/
!
!
voice translation-profile NOPLUS-IN
translate called 60
!
voice translation-profile PSTN-IN
translate calling 10
translate called 20
!
voice translation-profile PSTN-OUT
translate calling 30
translate called 40
!
voice translation-profile SRST-IN
translate called 50
!
!
!
license udi pid CISCO2951/K9 sn FJC2020A0PU
license accept end user agreement
license boot module c2951 technology-package uck9
hw-module pvdm 0/0
!
hw-module sm 1
!
!
!
username cbnetsvc password 7 08707D6F334B12040A
!
redundancy
!
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
description UPLINK TO DMZ-SW
ip address 10.46.30.5 233.233.233.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
description UPLINK TO SIP SERVICE
ip address 10.74.3.250 233.233.233.252
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/0/0
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 10.46.30.1
ip route 176.16.0.102 233.233.233.233 10.24.3.249
ip route 10.24.3.249 233.233.233.233 GigabitEthernet0/1
ip tacacs source-interface GigabitEthernet0/0
ip ssh time-out 60
ip ssh version 2
!
ip access-list standard SNMP_SERVERS
permit 176.17.240.170
permit 176.15.18.16
permit 176.17.104.5
permit 176.15.18.17
permit 176.15.18.15
permit 176.17.241.197
permit 176.17.223.27
permit 176.17.226.14
permit 176.17.240.8
permit 176.17.240.76
permit 176.17.241.68
permit 176.17.241.67
permit 176.17.198.80 0.0.0.15
deny any log
!
ip access-list extended BOX
permit ip any 45.58.64.0 0.0.15.233
permit ip 45.58.64.0 0.0.15.233 any
ip access-list extended DROPBOX
permit ip any 72.21.80.0 0.0.15.233
permit ip 72.21.80.0 0.0.15.233 any
ip access-list extended PCAP_ACL
permit ip any any
ip access-list extended WIRELESS-DATA
permit udp any any eq 5247
permit udp any eq 5247 any
!
logging trap notifications
logging source-interface GigabitEthernet0/0
logging host 176.17.206.60
logging host 176.17.105.85
!
nls resp-timeout 1
cpd cr-id 1
!
snmp-server community h0tsauce RO SNMP_SERVERS
snmp-server community gr33n RO SNMP_SERVERS
snmp-server community ch1ll1 RW SNMP_SERVERS
snmp-server ifindex persist
snmp-server location XPTO, Brazil
snmp-server system-shutdown
snmp-server enable traps tty
snmp-server enable traps syslog
tacacs-server host 176.17.198.120
tacacs-server host 176.15.18.13
tacacs-server directed-request
tacacs-server key 7 111D1A061E1406
!
!
!
control-plane
!
!
!
!
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm 192.168.104.1 identifier 2 version 7.0
sccp ccm 192.168.104.2 identifier 1 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register BRA_BR500_CFB
associate profile 2 register BRA_BR500_XCODE
associate profile 3 register BR500_711U_MTP
associate profile 4 register BR500_711A_MTP
associate profile 5 register BR500_729R8_MTP
keepalive retries 5
switchover method immediate
switchback method immediate
switchback interval 15
!
!
!
dspfarm profile 2 transcode
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g729r8
codec g729br8
maximum sessions 26
associate application SCCP
!
dspfarm profile 1 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 28
associate application SCCP
!
dspfarm profile 3 mtp
codec g711ulaw
maximum sessions software 100
associate application SCCP
!
dspfarm profile 4 mtp
codec g711alaw
maximum sessions software 100
associate application SCCP
!
dspfarm profile 5 mtp
codec g729r8
maximum sessions software 100
associate application SCCP
!
dial-peer voice 150 voip
preference 1
modem passthrough nse codec g711ulaw
session protocol sipv2
session target ipv4:192.168.104.2
destination e164-pattern-map 1
voice-class codec 1
dtmf-relay rtp-nte
fax rate 14400
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 250 voip
preference 2
modem passthrough nse codec g711ulaw
session protocol sipv2
session target ipv4:192.168.104.1
destination e164-pattern-map 1
voice-class codec 1
dtmf-relay rtp-nte
fax rate 14400
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 9911 voip
description Outbound calls to PSTN
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
preference 1
destination-pattern 0911$
session protocol sipv2
session target ipv4:176.16.0.102
incoming called e164-pattern-map 2
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 100 pots
tone ringback alert-no-PI
description Inbound calls from PSTN
incoming called-number .
direct-inward-dial
!
dial-peer voice 9112 voip
description Outbound calls to PSTN
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
preference 1
destination-pattern 0112$
session protocol sipv2
session target ipv4:176.16.0.102
incoming called e164-pattern-map 2
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 9192 voip
description Outbound calls to PSTN
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
preference 1
destination-pattern 019.$
session protocol sipv2
session target ipv4:176.16.0.102
incoming called e164-pattern-map 2
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 9181 voip
description Outbound calls to PSTN
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
preference 1
destination-pattern 0181$
session protocol sipv2
session target ipv4:176.16.0.102
incoming called e164-pattern-map 2
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 1000 voip
description Outbound calls from UCM
translation-profile incoming NOPLUS-IN
session protocol sipv2
incoming calling e164-pattern-map 1
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 110 voip
description Outbound calls to PSTN
translation-profile outgoing PSTN-OUT
destination-pattern 0T
session protocol sipv2
session target ipv4:176.16.0.102
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
voice-class sip profiles 1
voice-class sip profiles 1 inbound
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 1001 voip
description Inbound calls from CARRIER
translation-profile incoming PSTN-IN
session protocol sipv2
incoming called e164-pattern-map 2
voice-class codec 1
voice-class sip profiles 1
voice-class sip profiles 1 inbound
dtmf-relay rtp-nte
no vad
!
!
sip-ua
retry invite 2
timers trying 300
g729-annexb override
!
!
!
gatekeeper
shutdown
!
!
call-manager-fallback
max-conferences 16 gain -6
transfer-system full-consult
ip source-address 10.96.30.5 port 2000
max-ephones 75
max-dn 150 dual-line
system message primary WAN Outage - Fallback Mode
translation-profile incoming SRST-IN
!

XPTO_BR500_VGW#
XPTO_BR500_VGW#
XPTO_BR500_VGW#

Jaca · ‎04-20-2018

You have translation profiles on your dial peers.

You may want to do a show voice call stat to see what your in/out dial peers are and check them against these dial peers:

dial-peer voice 9112 voip
dial-peer voice 9192 voip
dial-peer voice 9181 voip
dial-peer voice 110 voip

Which all have your xlate profile outgoing hitting rule 30 and 40

From your config and without knowing your dialed number this would be a good starting point:

trunk group PSTN
translation-profile outgoing PSTN-OUT

!

voice translation-profile PSTN-OUT
translate calling 30
translate called 40

!

voice translation-rule 30
rule 2 /.*/ /8500/
rule 3 /^\+.*/ /8500/
!
voice translation-rule 40
rule 2 /^0$[2-9].......$$/ /\1/
rule 3 /^0$[79]........$$/ /\1/
rule 4 /^0$19.$$/ /\1/
rule 5 /^0$181$$/ /\1/
rule 6 /^0$911$$/ /\1/
rule 7 /^0$112$$/ /\1/
rule 8 /^00$[1-9][1-9][2-9].......$$/ /076\1/
rule 9 /^00$[1-9][1-9][2-9]........$$/ /076\1/
rule 10 /^00$[4-6][1-9]9[6-9].......$$/ /076\1/
rule 11 /^00$[1-37-9][1-9][79]........$$/ /076\1/
rule 12 /^00$[38]00.......$$/ /0\1/
rule 13 /^000$[1-9].*$$/ /0076\1/
rule 14 /^0$.*$$/ /\1/
!

On these dial peers:

dial-peer voice 9112 voip
dial-peer voice 9192 voip
dial-peer voice 9181 voip
dial-peer voice 110 voip

Jaca · ‎04-20-2018

You can also test your translation rule like such,

test voice translation rule 40 7408881212

and you'll get something back like:

pbx-router#test voice translation-rule 40 7408881212

Matched with rule 12

Original number: 2694003 Translated number: 0117408881212

Original number type: none Translated number type: none

Original number plan: none Translated number plan: none

Here's Cisco documentation on testing voice translation rules:

https://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/61083-voice-transla-rules.html

wilsonsant · ‎04-26-2018

Hi Jaca,

Sorry by long delay in answer. Unfortunatelly my profile don´t privilegius to run the command: test voice-translation-rule x

Thanks,

Wilson